rtmp2rtsp support G711U

pkg/base/avpacket.go

@@ -21,6 +21,7 @@ type AvPacketPt int
 
 const (
 	AvPacketPtUnknown AvPacketPt = -1
+	AvPacketPtG711U   AvPacketPt = 0  // g711u
 	AvPacketPtG711A   AvPacketPt = 8  // g711a
 	AvPacketPtAvc     AvPacketPt = 96 // h264
 	AvPacketPtHevc    AvPacketPt = 98 // h265

pkg/base/t_http_an__.go

@@ -15,7 +15,8 @@ package base
 
 const (
 	// AudioCodecAac StatGroup.AudioCodec
-	AudioCodecAac = "AAC"
+	AudioCodecAac   = "AAC"
+	AudioCodecG711U = "PCMU"
 
 	// VideoCodecAvc StatGroup.VideoCodec
 	VideoCodecAvc  = "H264"

pkg/base/t_rtmp.go

@@ -11,6 +11,7 @@ package base
 import (
 	"encoding/hex"
 	"fmt"
+
 	"github.com/q191201771/naza/pkg/bele"
 	"github.com/q191201771/naza/pkg/nazabytes"
 )
@@ -86,9 +87,12 @@ const (
 	//   AACAUDIODATA
 	//     AACPacketType UI8
 	//     Data          UI8[n]
-	RtmpSoundFormatAac         uint8 = 10 // 注意，视频的CodecId是后4位，音频是前4位
+	// 注意，视频的CodecId是后4位，音频是前4位
-	RtmpAacPacketTypeSeqHeader       = 0
+	RtmpSoundFormatG711U uint8 = 8
-	RtmpAacPacketTypeRaw             = 1
+	RtmpSoundFormatAac   uint8 = 10
+
+	RtmpAacPacketTypeSeqHeader = 0
+	RtmpAacPacketTypeRaw       = 1
 )
 
 type RtmpHeader struct {
@@ -129,13 +133,17 @@ func (msg RtmpMsg) IsVideoKeyNalu() bool {
 }
 
 func (msg RtmpMsg) IsAacSeqHeader() bool {
-	return msg.Header.MsgTypeId == RtmpTypeIdAudio && (msg.Payload[0]>>4) == RtmpSoundFormatAac && msg.Payload[1] == RtmpAacPacketTypeSeqHeader
+	return msg.Header.MsgTypeId == RtmpTypeIdAudio && msg.AudioCodecId() == RtmpSoundFormatAac && msg.Payload[1] == RtmpAacPacketTypeSeqHeader
 }
 
 func (msg RtmpMsg) VideoCodecId() uint8 {
 	return msg.Payload[0] & 0xF
 }
 
+func (msg RtmpMsg) AudioCodecId() uint8 {
+	return msg.Payload[0] >> 4
+}
+
 func (msg RtmpMsg) Clone() (ret RtmpMsg) {
 	ret.Header = msg.Header
 	ret.Payload = make([]byte, len(msg.Payload))

pkg/logic/group__core_streaming.go

@@ -394,8 +394,15 @@ func (group *Group) broadcastByRtmpMsg(msg base.RtmpMsg) {
 
 	// # 记录stat
 	if group.stat.AudioCodec == "" {
-		if msg.IsAacSeqHeader() {
+		if msg.Header.MsgTypeId == base.RtmpTypeIdAudio {
-			group.stat.AudioCodec = base.AudioCodecAac
+			switch msg.AudioCodecId() {
+			case base.RtmpSoundFormatAac:
+				if msg.IsAacSeqHeader() {
+					group.stat.AudioCodec = base.AudioCodecAac
+				}
+			case base.RtmpSoundFormatG711U:
+				group.stat.AudioCodec = base.AudioCodecG711U
+			}
 		}
 	}
 	if group.stat.VideoCodec == "" {

pkg/remux/rtmp2rtsp.go

@@ -10,10 +10,11 @@ package remux
 
 import (
 	"encoding/hex"
-	"github.com/q191201771/lal/pkg/h2645"
 	"math/rand"
 	"time"
 
+	"github.com/q191201771/lal/pkg/h2645"
+
 	"github.com/q191201771/lal/pkg/aac"
 	"github.com/q191201771/lal/pkg/avc"
 	"github.com/q191201771/lal/pkg/base"
@@ -74,6 +75,12 @@ func (r *Rtmp2RtspRemuxer) FeedRtmpMsg(msg base.RtmpMsg) {
 			Log.Warnf("rtmp msg too short, ignore. header=%+v, payload=%s", msg.Header, hex.Dump(msg.Payload))
 			return
 		}
+		if r.audioPt == base.AvPacketPtUnknown {
+			switch msg.AudioCodecId() {
+			case base.RtmpSoundFormatG711U:
+				r.audioPt = base.AvPacketPtG711U
+			}
+		}
 	case base.RtmpTypeIdVideo:
 		if len(msg.Payload) <= 5 {
 			Log.Warnf("rtmp msg too short, ignore. header=%+v, payload=%s", msg.Header, hex.Dump(msg.Payload))
@@ -136,7 +143,7 @@ func (r *Rtmp2RtspRemuxer) doAnalyze() {
 		}
 
 		// 回调sdp
-		ctx, err := sdp.Pack(r.vps, r.sps, r.pps, r.asc)
+		ctx, err := sdp.Pack(r.vps, r.sps, r.pps, r.asc, r.audioPt)
 		Log.Assert(nil, err)
 		r.onSdp(ctx)
 
@@ -206,26 +213,32 @@ func (r *Rtmp2RtspRemuxer) remux(msg base.RtmpMsg) {
 }
 
 func (r *Rtmp2RtspRemuxer) getAudioPacker() *rtprtcp.RtpPacker {
-	if r.asc == nil {
-		return nil
-	}
-
 	if r.audioPacker == nil {
 		// TODO(chef): ssrc随机产生，并且整个lal没有在setup信令中传递ssrc
 		r.audioSsrc = rand.Uint32()
 
-		ascCtx, err := aac.NewAscContext(r.asc)
+		switch r.audioPt {
-		if err != nil {
+		case base.AvPacketPtG711U:
-			Log.Errorf("parse asc failed. err=%+v", err)
+			pp := rtprtcp.NewRtpPackerPayloadPcm()
-			return nil
+			r.audioPacker = rtprtcp.NewRtpPacker(pp, 8000, r.audioSsrc)
-		}
+		case base.AvPacketPtAac:
-		clockRate, err := ascCtx.GetSamplingFrequency()
+			if r.asc == nil {
-		if err != nil {
+				return nil
-			Log.Errorf("get sampling frequency failed. err=%+v, asc=%s", err, hex.Dump(r.asc))
+			}
-		}
 
-		pp := rtprtcp.NewRtpPackerPayloadAac()
+			ascCtx, err := aac.NewAscContext(r.asc)
-		r.audioPacker = rtprtcp.NewRtpPacker(pp, clockRate, r.audioSsrc)
+			if err != nil {
+				Log.Errorf("parse asc failed. err=%+v", err)
+				return nil
+			}
+			clockRate, err := ascCtx.GetSamplingFrequency()
+			if err != nil {
+				Log.Errorf("get sampling frequency failed. err=%+v, asc=%s", err, hex.Dump(r.asc))
+			}
+
+			pp := rtprtcp.NewRtpPackerPayloadAac()
+			r.audioPacker = rtprtcp.NewRtpPacker(pp, clockRate, r.audioSsrc)
+		}
 	}
 	return r.audioPacker
 }

pkg/rtprtcp/rtp_packer_payload_pcm.go

@@ -0,0 +1,31 @@
+// Copyright 2021, Chef.  All rights reserved.
+// https://github.com/q191201771/lal
+//
+// Use of this source code is governed by a MIT-style license
+// that can be found in the License file.
+//
+// Author: Chef (191201771@qq.com)
+
+package rtprtcp
+
+type RtpPackerPayloadPcm struct {
+}
+
+func NewRtpPackerPayloadPcm() *RtpPackerPayloadPcm {
+	return &RtpPackerPayloadPcm{}
+}
+
+func (r *RtpPackerPayloadPcm) Pack(in []byte, maxSize int) (out [][]byte) {
+	if in == nil || maxSize <= 0 {
+		return
+	}
+
+	if len(in) > maxSize {
+		Log.Warnf("frame size bigger than rtp payload size while packing. len(in)=%d, maxSize=%d", len(in), maxSize)
+	}
+
+	item := make([]byte, len(in))
+	copy(item, in)
+	out = append(out, item)
+	return
+}

pkg/sdp/pack.go

@@ -21,17 +21,38 @@ import (
 	"github.com/q191201771/lal/pkg/base"
 )
 
-func Pack(vps, sps, pps, asc []byte) (ctx LogicContext, err error) {
+func Pack(vps, sps, pps, asc []byte, audioPt base.AvPacketPt) (ctx LogicContext, err error) {
 	// 判断音频、视频是否存在，以及视频是H264还是H265
-	var hasAudio, hasVideo, isHevc bool
+	var hasAudio, hasVideo, isHevc, isAac bool
 	if sps != nil && pps != nil {
 		hasVideo = true
 		if vps != nil {
 			isHevc = true
 		}
 	}
-	if asc != nil {
+
-		hasAudio = true
+	var samplingFrequency int
+	if audioPt != base.AvPacketPtUnknown {
+		switch audioPt {
+		case base.AvPacketPtG711U:
+			hasAudio = true
+			samplingFrequency = 8000
+		case base.AvPacketPtAac:
+			if asc != nil {
+				isAac = true
+				hasAudio = true
+				// 判断AAC的采样率
+				var ascCtx *aac.AscContext
+				ascCtx, err = aac.NewAscContext(asc)
+				if err != nil {
+					return
+				}
+				samplingFrequency, err = ascCtx.GetSamplingFrequency()
+				if err != nil {
+					return
+				}
+			}
+		}
 	}
 
 	if !hasAudio && !hasVideo {
@@ -39,20 +60,6 @@ func Pack(vps, sps, pps, asc []byte) (ctx LogicContext, err error) {
 		return
 	}
 
-	// 判断AAC的采样率
-	var samplingFrequency int
-	if asc != nil {
-		var ascCtx *aac.AscContext
-		ascCtx, err = aac.NewAscContext(asc)
-		if err != nil {
-			return
-		}
-		samplingFrequency, err = ascCtx.GetSamplingFrequency()
-		if err != nil {
-			return
-		}
-	}
-
 	sdpStr := fmt.Sprintf(`v=0
 o=- 0 0 IN IP4 127.0.0.1
 s=No Name
@@ -84,13 +91,21 @@ a=control:streamid=%d
 	}
 
 	if hasAudio {
-		tmpl := `m=audio 0 RTP/AVP 97
+		if isAac {
+			tmpl := `m=audio 0 RTP/AVP 97
 b=AS:128
 a=rtpmap:97 MPEG4-GENERIC/%d/2
 a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; config=%s
 a=control:streamid=%d
 `
-		sdpStr += fmt.Sprintf(tmpl, samplingFrequency, hex.EncodeToString(asc), streamid)
+			sdpStr += fmt.Sprintf(tmpl, samplingFrequency, hex.EncodeToString(asc), streamid)
+		} else {
+			tmpl := `m=audio 0 RTP/AVP 0
+a=rtpmap:0 PCMU/%d
+a=control:streamid=%d
+`
+			sdpStr += fmt.Sprintf(tmpl, samplingFrequency, streamid)
+		}
 	}
 
 	raw := []byte(strings.ReplaceAll(sdpStr, "\n", "\r\n"))

pkg/sdp/parse_logic.go

@@ -129,6 +129,8 @@ func ParseSdp2LogicContext(b []byte) (LogicContext, error) {
 				// 例子:a=rtpmap:8 PCMA/8000/1
 				// rtmpmap中有PCMA字段表示G711A
 				ret.audioPayloadTypeBase = base.AvPacketPtG711A
+			} else if strings.EqualFold(md.ARtpMap.EncodingName, ARtpMapEncodingNameG711U) {
+				ret.audioPayloadTypeBase = base.AvPacketPtG711U
 			} else {
 				if md.M.PT == 8 {
 					// ffmpeg推流情况下不会填充rtpmap字段,m中pt值为8也可以表示是PCMA,采样率默认为8000Hz

pkg/sdp/sdp.go

@@ -15,4 +15,5 @@ const (
 	ARtpMapEncodingNameH264  = "H264"
 	ARtpMapEncodingNameAac   = "MPEG4-GENERIC"
 	ARtpMapEncodingNameG711A = "PCMA"
+	ARtpMapEncodingNameG711U = "PCMU"
 )